The present invention relates to an adaptive prediction differential-PCM tranmission method and circuit. It is intended for use more particularly in telecommunications and particularly telephony.
The PCM technique (pulse-code modulation) is now widely used in the telecommunications field and in particular in telephone transmissions. This technique essentially comprises on transmission the sampling of the signal to be transmitted and then the quantization of the samples obtained, the coding of the quantized signals in digital form and finally the transmission of the coded signals, and on reception the decoding of the signal received, which supplies a sequence of samples from which the original signal is reconstituted.
At present, most International and National telephone networks use this technique. However, despite its considerable interest, it suffers from the disadvantage of necessitating a high information flow rate, which is of the order of 64 kbits/s (kilobits per second). Naturally, a considerable advance would be provided by reducing this flow rate, for example to 32 kbits/s, provided that the transmission quality was equal. However, this reduction is possible if account is taken of the intrinsic properties of the signal to be transmitted, which on this occasion is the speech signal.
Two categories of methods fulfilling this condition and which improve the PCM technique are known. These methods are called "differential coding" and "sub-bands". For a detailed description of these methods, reference should be made to the article by N. J. JAYANT, entitled "Digital Coding of Speech Waveforms: PCM, DPMC and DM Quantizers", published in the Journal "Proceedings of IEEE", May 1974, pp. 22 to 42. A brief description will be given of the principle of these two methods and thereby will facilitate understanding of the invention.
In the method using differential coding, the sampled input signal is not quantized, but instead the difference between this signal and a sample is predicted from the evolution of this difference.
The circuit diagram of a differential-PCM circuit is shown in FIG. 1. The part corresponding to the coding operation comprises an input 10 to which is applied a sampled signal Sn, a subtractor 12, having two inputs, one receiving the signal Sn and the other a predicted signal Sn, said subtracter having an output which supplies a differential signal dn, a quantizer 14 which supplies a quantized signal d'n which is then digitally coded by a coding circuit 16, an adder 18 which receives the quantized signal d'n and the predicted signal Sn and which supplies a signal S'n and a predictor circuit 20 which receives the signal S'n and supplies the predicted signal Sn
The part corresponding to the decoding operation comprises a decoding circuit 24 receiving the digital signal transmitted and supplying a signal d'n, an adder 26 having two inputs, one receiving the signal d'n and the other a predicted signal Sn, said adder supplying a signal S'n and a predictor circuit 28 receiving the signal S'n and supplying the predicted signal Sn.
In the diagram of FIG. 1 (as in the diagrams to be described hereinafter) the means for sampling the signal to be transmitted are not shown and it is assumed that the input signal is constituted by the sequence of samples Sn. In addition, the signal transmission and reception means and the signalling means are not shown, because they are well known in the art and do not form part of the present invention.
In the following description, the notations used in the first drawing will be retained, namely the use of a symbol n giving the order of the sampled signals, the letter d (optionally with an apostrophe) for the differential signals and the letter S (optionally with an apostrophe) for the other signals, the circumflex accent being used for marking the prediction signals.
The differential-PCM method is of particular interest when the differential signal dn is smaller than the input signal Sn, because then the quantization noise decreases compared with Sn and the signal/noise ratio improves. For this condition to be fulfilled, it is necessary that the predicted signal Sn is as close as possible to the input signal Sn, assuming that the latter is predictable.
This is precisely the case for the speech signal, particularly for human speech sounds, whose spectral and time evolutions are relatively slow. In general terms, the correlation between successive samples increases in inverse proportion to the dominant frequencies compared with the sampling frequency.
However, for the speech signal, the sampling frequency is generally equal to or above 8 kHz and the average longterm spectrum of speech has a maximum at around 500 Hz. Thus, conditions of good predictability exist.
In the differential PCM method, quantization can be carried out with a number of bits. For example, 4 bits are used for a sampling frequency of 8 kHz, corresponding to an information flow rate of 32 kBits/s. However, in a special variant called ".DELTA. coding", quantization is carried out with only a single bit for each sample, for which only the differential signal sign is used. However, as a counterpart to this reduction in the number of bits, the sampling frequency must be increased (and rises to 32 kHz if it is desired to retain the flow rate of 32 kbits/s).
In such a transmission system, the continuous modifications of the speech signal and the high dynamics of this signal lead to a considerable dispersion of the instantaneous levels occupied by differential signal dn. It is conventional practice to ensure a good adaptation between the high dispersion signal and the quantizer processing the same by multiplying the differential signal dn by a gain factor Gn in such a way that the signal produced dn.Gn which is processed by the quantizer, makes optimum use of the levels available therein. After quantization, the quantized signal is divided by the same factor Gn in order to reobtain the original quantized sample d'n. The gain factor Gn can be determined on the basis of an analysis of the levels occupied by the quantized signal d'n.Gn. This factor is reduced if d'n.Gn occupies too high levels and is reduced if d'n.Gn occupies too low levels.
A circuit diagram of a differential-PCM circuit provided with means for adapting the dynamics of the input signal is shown in FIG. 2 where the same elements as in FIG. 1 carry, for simplification purposes, the same references. In addition to the common elements, this circuit comprises firstly in the coding stage a multiplier 30 for the gain Gn which receives the differential signal dn, a circuit 32 able to calculate the gain Gn+1 necessary for the processing of the following sample and this calculation being based upon the signal d'n. Gn supplied by the quantizer 14, a circuit 34 for dividing by Gn placed between the quantizer 14 and an adder 18; and in the decoding stage a divider 36 by Gn inserted between the decoder 24 and the adder 26, and a circuit 38 for calculating the gain gn+1.
Circuits able to calculate the gain Gn+1 for a sample of order n+1 from the sample value of order n are well known to the those skilled in the art and will not be described here.
In a differential-PCM system, the prediction circuit, whose function is to supply a predicted signal Sn from a differential signal d'n, generally comprises a linear filter which, from a sequence of samples prior to the sample of order n, is able to give a prediction for the sample of order n.
Such a predictor filter can be calculated once and for all and is this case its characteristics are selected so that it is well adapted or matched to the long-term average spectrum of the signal to be transmitted. However, this type of filter does not make it possible to obtain a very good quality and in general 24 to 26 dB of the signal/noise ratio. For the 64 kbit/s PCM system, this ratio can reach 38 dB and must, in operation, always exceed 33.5 dB. It is known that such ratios can be measured by the isopreference method, which consists of alternately listening to coded phrases and the same phrases in the uncoded state on which are superimposed a multiplicative white noise, which is adjusted so that the same noise impression is obtained for both signals.
The transmission quality of a differential-PCM system can be improved by adapting the predictor filter to the evolutions of the signal in time, by a periodic updating of its characteristics. This adaptation can either be carried out on an overall basis or sequentially. It is carried out on an overall basis by calculating on sample blocks (chosen sufficiently short to enable the signal to be considered as stationary and in practice of the order of 20 ms) the filter most suitable for each signal band. This method has the double disadvantage of introducing a systematic delay corresponding to the length of the sample block used and of necessitating the transmission to the reception stage of information coefficients relative to the filter used in the transmission stage. This systematic delay and this supplementary transition mean that this method cannot be considered for modern telephony. The alternative is sequential adaptation by correcting at each sampling instant the characteristics of the filter in accordance with the differential signal value at this instant. This sequential method has the disadvantage of being difficult to perform, without any great resulting improvement over a fixed prediction (the gain is only 2 to 3 dB). Such a technique is described in the article by J. D. GIBSON entitled "Sequentially Adaptive Backward Predication in ADPCM Speech Coders", published in the Journal "IEEE Transactions on Communications", vol. COM-26, No. 1, January 1978.
As regards the second category of transmission methods, in which account is taken of the special properties of the signal to be transmitted, the coding is accomplished by sub-bands method. This consists of filtering the signal to be transmitted in adjacent band pass filters, the output of each filter being transposed in the corresponding base band, then filtering the signals corresponding to each sub-band with a frequency which is double the band width. In this connection, reference can be made to the article by R. E. CROCHIERE et al, entitled "Digital Coding of Speech in Sub-Bands", published in the Journal "The Bell System Technical Journal", October 1976, pp. 1069 to 1085.
Such a method has two advantages in that for each sub-band, the quantization noise remaining limited to said band is strictly correlated to the signal and is therefore subjectively only very slightly disturbing. Moreover, the information flow rate attributed to each band can be modified as a function of the evolution of energies in said band.
However, this method also has two disadvantages. The construction of band pass filters and the scaling of the quantizers imply significant delays (of the order of 12 to 16 ms) which are incompatible with use in a telephone system. Finally, this method requires the parallel transmission of the scaling coefficients of the quantizers, so that multiplexing is relatively complicated.